1. Field of the Invention
The present invention relates to telephone communications and more particularly to collection of past due balances from telephone subscribers.
2. Description of Related Art
The invention is particularly applicable in the context of internet telephony (also known as voice-over-IP (VoIP) or IP telephony), which should be understood to be a telephone system in which voice is communicated via a packet switched network or link (such as the Internet, or an ATM, X.25 or other packet-switched connection) between two or more parties. Internet telephony may also be referred to as packet switched telephony.
In general, an internet telephony system facilitates telephone communication between two or more users over a packet switched network, such as an IP network for example. Each user is positioned at a telephone device (hereafter “telephone”), which is generally any communications device capable of communicating real-time media signals such as speech, audio and/or video, for example.
By way of example and without limitation, the telephone device may be a conventional analog telephone, a digital telephone, a videophone, and/or a multi-media personal computer. Each telephone device (and/or telephone number) is then typically served by a local media gateway (or “internet telephony gateway”), which provides connectivity to the packet switched network. Alternatively, the telephone device itself may provide connectivity with the packet switched network and may serve other gateway-functions as well. Such a telephone device may be referred to as an “internet telephone” and may take any of a variety of forms now known or later developed.
To place a call over a packet switched network via an initiating gateway, a user at an initiating telephone device may establish a connection with the initiating gateway via a suitable communications link such as the public switched telephone network (PSTN), via a wireless access channel, and/or via some other circuit switched or packet switched network or direct link. And the user may specify a telephone number of the called party. The initiating gateway may then query an address mapping database or consult with another device or process in the system to determine a network address of a terminating gateway that serves the called number.
In turn, the initiating gateway may contact the terminating gateway via the packet switched network, notifying the terminating gateway of a desire to establish a connection with the called party. The terminating gateway may then establish an appropriate connection with a telephone device at the called number and notify the initiating gateway that the call can proceed. With the end-to-end connection thus established between the calling and called parties, the parties may then communicate with each other over the packet switched network.
In an internet telephony system, as in any telephony system, a signaling system must be provided in order to facilitate various functions involved in setting up calls. One such signaling system is defined by the Session Initiation Protocol (SIP), which is well known to those of ordinary skill in the art.
According to SIP, each signaling endpoint includes a user agent client (UAC) application, which facilitates SIP signaling. In order for a first endpoint to initiate communication with a second endpoint, the first endpoint sends to the IP address of the second endpoint a SIP “INVITE” message, specifying the type of communication desired. If the second endpoint agrees to communicate, the second endpoint then sends to the first endpoint a SIP “200 OK” message, signaling agreement. And the first endpoint in turn sends a SIP “ACK” message to the second endpoint, to complete the session setup. The two ends may then begin communicating with each other in the specified manner.
In practice, each user in a SIP-capable system will have a personal identifier called a SIP address, typically in the form “username@realm.” And the network will include a SIP registration server that correlates each user's SIP address with the IP address of the client station that the user is currently operating. When initiating a communication, the UAC on the initiating client device can then send a SIP INVITE to a local proxy server, indicating in the SIP INVITE (i) the SIP address of the calling party and (ii) the SIP address of the called party. And the proxy server can consult the SIP registration server to find out the IP address where the called party's SIP address is currently registered and can forward the SIP INVITE to that IP address. Alternatively, the initiating UAC could query a SIP redirect server to determine the destination IP address and can then itself send the SIP INVITE to that address rather than to the proxy server.
Oftentimes, a telephone device will not itself include a SIP UAC. In that event, the telephone device might instead engage in signaling communication with its local gateway through more conventional (legacy) signaling mechanisms. And the gateway might then include a UAC that can operate on behalf of the telephony device. For instance, the gateway could receive a legacy call initiation signal from the telephony device (or from a switch or other node serving the device) and could responsively generate and transmit a SIP INVITE seeking to set up the desired session over a packet-switched network. In any event, a gateway will typically either receive and send, or generate and send, a SIP INVITE on behalf of the user.
In an internet telephony system, a SIP INVITE can designate the called party by a SIP address that includes the called party's telephone number, such as sip:+1-913-555-1234@realm.com. If the called party is operating a SIP-capable telephone, then the SIP registration server may correlate that SIP address with the IP address of the telephone, so as to facilitate sending the INVITE to the called party's telephone. Alternatively, if the called party is operating a telephone that is not SIP-capable, then the SIP registration sever may correlate that SIP address with the IP address of a SIP-capable gateway serving the called party's telephone, so as to facilitate sending the INVITE to that gateway, and the gateway can then send a corresponding legacy signal to the called party's telephone.
An internet telephony system may also include a call server (or “application call server”) that functions to assist the gateway in setting up and conducting calls. The call server might be owned or operated by a carrier that provides telecommunication services for designated subscribers, typically billing those subscribers for the service.
When setting up a call for a user, a gateway might signal to the call server, and the call server might then query a backend system, such as an authentication/authorization/accounting (AAA) server, to determine whether or not to allow the call to go through. In particular, the AAA server could contain profile records for each authorized user/client. And, as is known in the art, the call server could query the AAA server to determine whether the caller's account is current and valid, such as to ensure that the caller has paid all past bills, or to verify the caller's personal identification number or security code. The call server can then send a response signal to the gateway, either allowing the requested call to proceed or not.
In a SIP-based signaling system, for instance, an initiating gateway could signal to the call server by simply sending the SIP INVITE to the call server instead of to a remote signaling endpoint such as a terminating gateway or the called telephone. The call server could then query the AAA server on behalf of the initiating gateway. If the AAA server authorizes the call to proceed, the call server could then send the SIP INVITE back to the gateway for transmission in turn to the remote signaling endpoint. Alternatively, if the AAA server determines that the call should not proceed, the call server could send a failure result message (such as a SIP 600 or 700 series message) to the gateway, to prevent the call from going through.